[asterisk-users] Trying to make SIP calls through Asterisk with anonymous connection

[Royce Souther]

I am trying to setup SIP to SIP calling between Asterisk managed networks. I
want to make it so that people can call SIP:123 at MyDomain.URL and they
connect to my Asterisk and get my external IVR then they can dial my
extension or navigate extensions just like they would if they had called
using a PSTN line. I also want to call other people using my Asterisk and
dialing an external SIP like so SIP:321 at SomeOther.URL. I want out going SIP
calls to me managed by my Asterisk so I can transfer them to other people in
my office or conference or use any of the other great features that Asterisk
provides. I do not want to go SIP direct to SIP, I want to go SIP to
Asterisk to Asterisk to SIP and connect to the far end Asterisk without
requiring me to register my Asterisk server with the far end Asterisk
server.

For testing I have setup two servers running Asterisk. Both are on the
Internet with static IP addresses and behind firewalls. The firewalls are
configured to allow TCP & UDP ports 5060 to 5082 and 10001 to 20000 to
connect directly to the Asterisk servers. This allows SIP and RTP
connections from the outside. I have tested with Twinkle (a Linux softphone)
and can connect to a registered account with NAT from external IPs. I have
also set the Asterisk servers to allow incoming anonymous SIP calls to
connect to the from-external.  When I try to dial SIP:some_extension at other_url
Asterisk tries to dial the some_extension on my local network not the other
network. I reconnect to the running asterisk using -r and watch when I dial
and it does not report the @other_url only the some_extension.

I am not having much luck finding the documentation I need. Can someone
point me to a How-To on doing this?

-- 
Open Source: To innovate then create
Proprietary: To imitate then litigate
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